Windows Vista/7¶

• Press the start button and type 'ncpa.cpl' into the search box and press enter. This will open the 'network connections' window.
• Press Alt-N to open the hidden 'Advanced' menu and choose 'Advanced Settings'.
• Adjust the order of the network cards listed in the first box marked 'Connections' and press OK when done.
  

Windows XP¶

• Press the start button and select 'Run' - type 'ncpa.cpl' and press enter. This will open the 'network connections' window.
• Open the hidden 'Advanced' menu and choose 'Advanced Settings'.
• Adjust the order of the network cards listed in the first box marked 'Connections' and press OK when done.
  

Downloading the software¶

Starting Solo for the first time¶

Audio Settings¶

ASIO¶

Configure inputs/outputs¶

SIP Providers¶

SIP settings¶

Display¶

• Lines - The number of lines available on the gateway/with the provider. This will directly control how many calls can be made and received in Solo from the provider.
• Dial priority - When clicking the dial button in Solo a provider is chosen to dial on using the dial priority. Providers with a lower number will be tried in turn until a provider is found with spare lines.
• Line label - The line label will be shown on each line for the provider. It is a good place to enter a description (e.g. XD) or the actual phone number of the line.
  

SIP Server¶

• Registrar/Proxy - The IP address and port of the gateway/provider.
• User @ domain - The user name and domain name required for authentication to connect to the gateway/provider.
• Auth id - (Optional) If the auth id is different to the user name for the provider/gateway specify it here.
• Password - If a password is required by the gateway/provider enter it here.
• Registration - tick register if the gateway/provider supports registration, and specify an interval that registration should occur.
• Allow direct calls - if ticked, calls to alternate SIP domains will be made directly from Solo, not via the gateway.
• Local IP address - leave set to '(auto)' except for advanced IP configurations involving firewalls and port forwarding.
  

NAT/Firewall¶

• STUN Server - Set the address of a STUN server, if known. STUN allows discovery of paths through NAT firewalls. Not usually required - set this value only if inbound calls or their audio is not received when using a SIP provider on the Internet. STUN should only be used a a last resort method of negotiating NAT and in most cases in not required.
• 'Keep alive' interval - To keep NAT translation active a keep alive can be sent on a regular basis.
  

Audio¶

• Max jitter (ms) - The maximum jitter controls the variation in interval of audio packets received over IP. If the jitter is too small glitches in audio may be heard. If the jitter is set too high, the latency of the call will be excessive.
  

Number rules¶

• Dial prefix - a number to insert in front of any dialled number, e.g. 9 for an outside line. This would then not need to be included in any dialled number within Solo. The minimum digits controls when the prefix will be applied. E.g. numbers with less than 5 digits are internal calls - don't apply the 9.
• Strip prefix - a number to strip from the front of received calls. If a gateway/provider is prefixing calls with an unrequired digit specify it here to remove it. The minimum digits controls when the prefix will be applied.
• Strip characters - any characters specified here will be removed from any number (dialled or received). E.g. enter ',-' to have commas and dashes removed from numbers if caller id is presenting numbers which contain these characters.
• Strip local area code when dialling - if the provider cannot dial local numbers when the local area code is included in the dialled number, select this option to remove the area code before passing it to the provider.
  

Using a handset¶

Configuring Solo¶

Configuring the handset (Linksys SPA9xx example using web based configuration tool)¶

Using GPIO cards¶

• For Analogue outputs - Select the ‘A 1-2’ button on the right of the control panel to show the inputs which are currently being fed into the analogue output mix. Ensure that the ‘m’ buttons for ‘analog in’ are depressed (analog in ‘m’ buttons' circled in screen shot).
• For AES outputs - instead, select the ‘D 1-2’ button on the right of the control panel to show the inputs which are currently being fed into the digital output mix. Ensure that the ‘m’ buttons for ‘digital in’ are depressed.
  

Introduction¶

Requirements¶

Main Solo¶

• A SIP gateway or service. Solo will register with the gateway to make and receive calls.
• A Windows PC to run Solo on.
• A supported soundcard. With Solo 2.0 and above any WAVE or ASIO soundcard should work – but see our other tips for more information on this as better cards have lower latency.
• Optional – a Linksys handset to screen callers without using the mixing desk.
• Optional - Advantech GPI/O card for service ringing/device occupied outputs and answer and drop inputs.
  

Buddy¶

• A Windows PC to run Solo on.
• For a buddy with audio – a supported soundcard.
• For a buddy without audio – you will simply be controlling the calls on the main Solo.
• Optional – a Linksys handset to screen callers without using the mixing desk.
  

Minimum specification¶

• Microsoft Windows XP or above
• 2Ghz or faster processor
• 1GB RAM
• 10GB Hard disk space
• 1024x768 resolution
  

Solo Topology¶

Using a SIP provider¶

Using ISDN lines and a SIP gateway¶

Using Analogue lines and a SIP gateway¶

Audio to sound desk¶